Error correction for DTMF corruption on uplink

ABSTRACT

Aspects relate to provision of enterprise call capabilities to mobile devices. For example, a mobile device can indicate, over a data channel, that a PBX is to make a call on its behalf to a called party. The PBX can call back the mobile device, call the called party, and bridge those call legs to establish the call. The mobile device can employ mechanisms that a particular incoming call is made by the PBX. These mechanisms can include using ANI information, sending, and receiving audible verification codes over the voice channel established after answering the incoming call. The verification codes can be selected based different behaviors of the mobile devices.

CROSS-REFERENCE TO RELATED APPLICATION

This application is a continuation of U.S. application Ser. No.12/692,951, filed Jan. 25, 2010, the entire contents of which areincorporated by reference herein.

FIELD

The present application relates to voice telephony on mobile devices,and more particularly relates to call control and status updating fortelephony.

BACKGROUND

Voice telephony remains a major application of interest on mobiledevices, such as smartphones. Typically, mobile devices implement voicetelephony over circuits (similar to the public switch telephone network(PSTN)), once past the radio access network. In some cases, mobiledevices may support a data channel, in addition to a voice channel(e.g., such devices may support concurrent voice and datacommunications). In such situations, a service provider may perform atleast some voice call control functions over the data channel. However,even if a given device supports simultaneous voice and datacommunications, data communications may not be available on allnetworks, or may be sporadically unavailable, such that there may besituations where even though a voice call can be made, a data channel isunavailable.

BRIEF DESCRIPTION OF THE DRAWINGS

Reference will now be made, by way of example, to the accompanyingdrawings which show example embodiments of the present application, andin which:

FIG. 1 shows, in block diagram form, an example system for managingenterprise-related mobile calls, including an enterprise communicationsplatform;

FIG. 2 shows, in block diagram form, further details of an embodiment ofthe enterprise communications platform;

FIG. 3 shows another embodiment of the enterprise communicationsplatform;

FIG. 4 shows yet another embodiment of the enterprise communicationsplatform;

FIG. 5 shows further details of the enterprise communications platformof FIG. 3;

FIG. 6A is a signaling diagram generally indicating howmobile-originated, mobile-initiated calls are processed by the networkof FIG. 5;

FIG. 6B is a signaling diagram generally indicating howmobile-originated, PBX-initiated, calls are processed by the network ofFIG. 5;

FIG. 7A is a signaling diagram generally indicating howmobile-terminated, mobile-initiated calls are processed by the networkof FIG. 5;

FIG. 7B is a signaling diagram generally indicating howmobile-terminated, PBX-initiated calls are processed by the network ofFIG. 5;

FIG. 8 depicts example of components of an example mobile device;

FIG. 9 depicts an example form factor of a mobile device;

FIG. 10 depicts an example of functional modules that may be provided ina mobile device;

FIG. 11 depicts an abstraction of an example system for in progresscommand and status updates for a voice call;

FIG. 12 depicts more detail concerning a module of the system of FIG.11.

FIG. 13 depicts an example of state-dependent DTMF code translation;

FIG. 14 depicts a method in which a mobile device can participate; and

FIG. 15 depicts a method in which a PBX or server that is handling acall with a mobile device can participate.

DESCRIPTION

In general, the present application relates to the control andmanagement of communications. In one exemplary aspect, the presentapplication relates to a telephony method for implementation on a mobiledevice. The present disclosure includes call control and call statussharing techniques in the absence of a data channel. The telephonymethod comprises receiving an incoming voice call over a voice channelon the mobile device. The incoming voice call may be from a PBX (or moregenerally, a platform providing enterprise communication capabilities tomobile devices, for simplicity these platforms are referred herein as a“PBX”). The method includes answering the voice call and sending, fromthe mobile device, a verification code comprising a series of audibletones, when identifying information for the voice call being received isunavailable to the mobile device. The method allows the voice call toproceed responsive to receiving, at the mobile device, a verificationcode over the voice channel within a time limit. The method can beemployed, for example, where the mobile device has signaled to a PBXthat it wants the PBX to make a call on its behalf. The PBX can make thecall, and when the called party has accepted the call, the PBX can callthe mobile device, and bridge both call legs, establishing the call.These example is by way of explanation, rather than limitation.

Although reference may be made to “calls” in the description of exampleembodiments below, it will be appreciated that the described systems andmethods are applicable to session-based communications in general andnot limited to voice calls. Other aspects of the present applicationwill be apparent to those of ordinary skill in the art from a review ofthe following detailed description in conjunction with the drawings.Embodiments of the present application are not limited to any particularoperating system, mobile device architecture, server architecture, orcomputer programming language.

Reference is now made to FIG. 1, which shows, in block diagram form, anexample system, generally designated 10, for the control and managementof communications. The system 10 includes an enterprise or businesssystem 20, which in many embodiments includes a local area network(LAN). In the description below, the enterprise or business system 20may be referred to as an enterprise network 20. It will be appreciatedthat the enterprise network 20 may include more than one network and maybe located in multiple geographic areas in some embodiments.

The enterprise network 20 may be connected, often through a firewall 22,to a wide area network (WAN) 30, such as the Internet. The enterprisenetwork 20 may also be connected to a public switched telephone network(PSTN) 40 via direct inward dialing (DID) trunks or primary rateinterface (PRI) trunks.

The enterprise network 20 may also communicate with a public land mobilenetwork (PLMN) 50, which may also be referred to as a wireless wide areanetwork (WWAN) or, in some cases, a cellular network. The connectionwith the PLMN 50 may be made via a relay 26.

The enterprise network 20 may also provide a wireless local area network(WLAN) 32 a featuring wireless access points. Other WLANs 32 may existoutside the enterprise network 20. For example, WLAN 32 b may beconnected to WAN 30.

The system 10 may include a number of enterprise-associated mobiledevices 11 (only one shown). The mobile devices 11 may include devicesequipped for cellular communication through the PLMN 50, mobile devicesequipped for Wi-Fi communications over one of the WLANs 32, or dual-modedevices capable of both cellular and WLAN communications. WLANs 32 maybe configured in accordance with one of the IEEE 802.11 specifications.

It will be understood that the mobile devices 11 include one or moreradio transceivers and associated processing hardware and software toenable wireless communications with the PLMN 50 and/or one of the WLANs32. In various embodiments, the PLMN 50 and mobile devices 11 may beconfigured to operate in compliance with any one or more of a number ofwireless protocols, including GSM, GPRS, CDMA, EDGE, UMTS, EvDO, HSPA,3GPP, or a variety of others. It will be appreciated that the mobiledevice 11 may roam within the PLMN 50 and across PLMNs, in known manner,as the user moves. In some instances, the dual-mode mobile devices 11and/or the enterprise network 20 are configured to facilitate roamingbetween the PLMN 50 and a WLAN 32, and are thus capable of seamlesslytransferring sessions (such as voice calls) from a connection with thecellular interface of the dual-mode device 11 to the WLAN 32 interfaceof the dual-mode device 11, and vice versa.

The enterprise network 20 typically includes a number of networkedservers, computers, and other devices. For example, the enterprisenetwork 20 may connect one or more desktop or laptop computers 15 (oneshown). The connection may be wired or wireless in some embodiments. Theenterprise network 20 may also connect to one or more digital telephonesets 17 (one shown).

The enterprise network 20 may include one or more mail servers, such asmail server 24, for coordinating the transmission, storage, and receiptof electronic messages for client devices operating within theenterprise network 20. Typical mail servers include the MicrosoftExchange Server™ and the IBM Lotus Domino™ server. Each user within theenterprise typically has at least one user account within the enterprisenetwork 20. Associated with each user account is message addressinformation, such as an e-mail address. Messages addressed to a usermessage address are stored on the enterprise network 20 in the mailserver 24. The messages may be retrieved by the user using a messagingapplication, such as an e-mail client application. The messagingapplication may be operating on a user's computer 15 connected to theenterprise network 20 within the enterprise. In some embodiments, theuser may be permitted to access stored messages using a remote computer,for example at another location via the WAN 30 using a VPN connection.Using the messaging application, the user may also compose and sendmessages addressed to others, within or outside the enterprise network20. The messaging application causes the mail server 24 to send acomposed message to the addressee, often via the WAN 30.

The relay 26 serves to route messages received over the PLMN 50 from themobile device 11 to the corresponding enterprise network 20. The relay26 also pushes messages from the enterprise network 20 to the mobiledevice 11 via the PLMN 50.

The enterprise network 20 also includes an enterprise server 12.Together with the relay 26, the enterprise server 12 functions toredirect or relay incoming e-mail messages addressed to a user's e-mailaddress within the enterprise network 20 to the user's mobile device 11and to relay incoming e-mail messages composed and sent via the mobiledevice 11 out to the intended recipients within the WAN 30 or elsewhere.The enterprise server 12 and relay 26 together facilitate “push” e-mailservice for the mobile device 11 enabling the user to send and receivee-mail messages using the mobile device 11 as though the user wereconnected to an e-mail client within the enterprise network 20 using theuser's enterprise-related e-mail address, for example on computer 15.

As is typical in many enterprises, the enterprise network 20 includes aPrivate Branch eXchange (although in various embodiments the PBX may bea standard PBX or an IP-PBX, for simplicity the description below usesthe term PBX to refer to both) 16 having a connection with the PSTN 40for routing incoming and outgoing voice calls for the enterprise. ThePBX 16 is connected to the PSTN 40 via DID trunks or PRI trunks, forexample. The PBX 16 may use ISDN signaling protocols for setting up andtearing down circuit-switched connections through the PSTN 40 andrelated signaling and communications. In some embodiments, the PBX 16may be connected to one or more conventional analog telephones 19. ThePBX 16 is also connected to the enterprise network 20 and, through it,to telephone terminal devices, such as digital telephone sets 17,softphones operating on computers 15, etc. Within the enterprise, eachindividual may have an associated extension number, sometimes referredto as a PNP (private numbering plan), or direct dial phone number. Callsoutgoing from the PBX 16 to the PSTN 40 or incoming from the PSTN 40 tothe PBX 16 are typically circuit-switched calls. Within the enterprise,e.g. between the PBX 16 and terminal devices, voice calls are oftenpacket-switched calls, for example Voice-over-IP (VoIP) calls.

The enterprise network 20 may further include a Service ManagementPlatform (SMP) 18 for performing some aspects of messaging or sessioncontrol, like call control and advanced call processing features. TheSMP 18 may, in some cases, also perform some media handling.Collectively the SMP 18 and PBX 16 may be referred to as the enterprisecommunications platform (server), generally designated 14. It will beappreciated that the enterprise communications platform 14 and, inparticular, the SMP 18, is implemented on one or more servers havingsuitable communications interfaces for connecting to and communicatingwith the PBX 16 and/or DID/PRI trunks. Although the SMP 18 may beimplemented on a stand-alone server, it will be appreciated that it maybe implemented into an existing control agent/server as a logicalsoftware component. As will be described below, the SMP 18 may beimplemented as a multi-layer platform.

The enterprise communications platform 14 implements the switching toconnect session legs and may provide the conversion between, forexample, a circuit-switched call and a VoIP call, or to connect legs ofother media sessions. In some embodiments, in the context of voice callsthe enterprise communications platform 14 provides a number ofadditional functions including automated attendant, interactive voiceresponse, call forwarding, voice mail, etc. It may also implementcertain usage restrictions on enterprise users, such as blockinginternational calls or 1-900 calls. In many embodiments, SessionInitiation Protocol (SIP) may be used to set-up, manage, and terminatemedia sessions for voice calls. Other protocols may also be employed bythe enterprise communications platform 14, for example, Web Services,Computer Telephony Integration (CTI) protocol, Session InitiationProtocol for Instant Messaging and Presence Leveraging Extensions(SIMPLE), and various custom Application Programming Interfaces (APIs),as will be described in greater detail below.

One of the functions of the enterprise communications platform 14 is toextend the features of enterprise telephony to the mobile devices 11.For example, the enterprise communications platform 14 may allow themobile device 11 to perform functions akin to those normally availableon a standard office telephone, such as the digital telephone set 17 oranalog telephone set 15. Example features may include direct extensiondialing, enterprise voice mail, conferencing, call transfer, call park,etc.

Reference is now made to FIGS. 2 to 4, which show example embodiments ofthe enterprise communications system 14. Again, although references aremade below to “calls” or call-centric features it will be appreciatedthat the architectures and systems depicted and described are applicableto session-based communications in general and, in some instances, tomessaging-based communications.

FIG. 2 illustrates an embodiment intended for use in a circuit-switchedTDM context. The PBX 16 is coupled to the SMP 18 via PRI connection 60or other suitable digital trunk. In some embodiments, the PRI connection60 may include a first PRI connection, a second PRI connection, and achannel service unit (CSU), wherein the CSU is a mechanism forconnecting computing devices to digital mediums in a manner that allowsfor the retiming and regeneration of incoming signals. It will beappreciated that there may be additional or alternative connectionsbetween the PBX 16 and the SMP 18.

In this embodiment, the SMP 18 assumes control over both call processingand the media itself. This architecture may be referred to as “FirstParty Call Control”. Many of the media handling functions normallyimplemented by the PBX 16 are handled by the SMP 18 in thisarchitecture. Incoming calls addressed to any extension or direct dialnumber within the enterprise, for example, are always first routed tothe SMP 18. Thereafter, a call leg is established from the SMP 18 to thecalled party within the enterprise, and the two legs are bridged.Accordingly, the SMP 18 includes a digital trunk interface 62 and adigital signal processing (DSP) conferencing bridge 64. The DSPconferencing bridge 64 performs the bridging of calls for implementationof various call features, such as conferencing, call transfer, etc. Thedigital trunk interface 62 may be implemented as a plurality oftelephonic cards, e.g. Intel Dialogic cards, interconnected by a bus andoperating under the control of a processor. The digital trunk interface62 may also be partly implemented using a processor module such as, forexample, a Host Media Processing (HMP) processor.

The SMP 18 may include various scripts 66 for managing call processing.The scripts 66 are implemented as software modules, routines, functions,etc., stored in non-volatile memory and executed by the processor of theSMP 18. The scripts 66 may implement call flow logic, business logic,user preferences, call service processes, and various featureapplications.

FIG. 3 shows another embodiment in which the PBX 16 performs thefunctions of terminating and/or bridging media streams, but call controlfunctions are largely handled by the SMP 18. In this embodiment, the SMP18 may be referred to as a call control server 18. This architecture maybe referred to as “Third-Party Call Control”.

The call control server 18 is coupled to the PBX 16, for example throughthe LAN, enabling packet-based communications and, more specifically,IP-based communications. In one embodiment, communications between thePBX 16 and the call control server 18 are carried out in accordance withSIP. In other words, the call control server 18 uses SIP-basedcommunications to manage the set up, tear down, and control of mediahandled by the PBX 16. In one example embodiment, the call controlserver 18 may employ a communications protocol conforming to theECMA-269 or ECMA-323 standards for Computer Supported TelecommunicationsApplications (CSTA).

FIG. 4 shows yet another embodiment of the enterprise communicationssystem 14. This embodiment reflects the adaptation of an existing set ofcall processing scripts to an architecture that relies on third-partycall control, with separate call control and media handling. The SMP 18includes a call processing server 74. The call processing server 74includes the scripts or other programming constructs for performing callhandling functions. The SMP 18 also includes a SIP server 72 and a mediaserver 76. The separate SIP server 72 and media server 76 logicallyseparate the call control from media handling. The SIP server 72interacts with the call processing server 74 using acomputer-implemented communications handling protocol, such as one ofthe ECMA-269 or ECMA-323 standards. These standards prescribe XML basedmessaging for implementing Computer Supported TelecommunicationsApplications (CSTA).

The SIP server 72 interacts with the media server 76 using SIP-basedmedia handling commands. For example, the SIP server 72 and media server76 may communicate using Media Server Markup Language (MSML) as definedin IETF document Saleem A., “Media Server Markup Language,” InternetDraft, draft-saleem-msml-07, Aug. 7, 2008. The media server 76 may beconfigured to perform Host Media Processing (HMP).

Other architectures or configurations for the enterprise communicationssystem 14 will be appreciated by those ordinarily skilled in the art.

Reference is now made to FIG. 5, which shows another embodiment of theenterprise communications system 14 with a Third Party Call Controlarchitecture. In this embodiment, the SMP 18 is a multi-layer platformthat includes a protocol layer 34, a services layer 36 and anapplication layer 38. The protocol layer 34 includes a plurality ofinterface protocols configured for enabling operation of correspondingapplications in the application layer 38. The services layer 36 includesa plurality of services that can be leveraged by the interface protocolsto create richer applications. Finally, the application layer 38includes a plurality of applications that are exposed out to thecommunication devices and that leverage corresponding ones of theservices and interface protocols for enabling the applications.

Specifically, the protocol layer 34 preferably includes protocols whichallow media to be controlled separate from data. For example, theprotocol layer 34 can include, among other things, a Session InitiationProtocol or SIP 80, a Web Services protocol 82, an ApplicationProgramming Interface or API 84, a Computer Telephony Integrationprotocol or CTI 86, and a Session Initiation Protocol for InstantMessaging and Presence Leveraging Extensions or SIMPLE protocol 88. Itis contemplated that the interface protocols 80-88 are plug-ins that caninterface directly with corresponding servers in the enterprise network20, which will be further described below.

For the purposes of this disclosure, SIP 80 will be utilized, althoughit is appreciated that the system 10 can operate using the abovedisclosed or additional protocols. As known by those of ordinary skillin the art, SIP is the IETF (Internet Engineering Task Force) standardfor multimedia session management, and more specifically is anapplication-layer control protocol for establishing, maintaining,modifying and terminating multimedia sessions between two or moreendpoints. As further known by those of ordinary skill in the art, theSIP protocol 80 includes two interfaces for signaling: SW-Trunk(hereinafter referred to as “SIP-T”) and SIP-Line (hereinafter referredto as “SIP-L”). Specifically, the SIP-T interface is utilized when theendpoint is a non-specific entity or not registered (i.e., whencommunicating between two network entities). In contrast, the SIP-Linterface is utilized when the endpoint is registered (i.e., whendialing to a specific extension). The specific operation of the system10 utilizing SIP 80 will be described in further detail below.

The SMP 18 also includes a plurality of enablers, among other things, aVoIP enabler 90, a Fixed Mobile Convergence or FMC enabler 92, aconference services enabler 94, a presence enabler 96 and an InstantMessaging or IM enabler 98. Each of the enablers 90-98 are used bycorresponding services in the services layer 36 that combine one or moreof the enablers. Each of the applications in the application layer 38 isthen combined with one or more of the services to perform the desiredapplication. For example, a phone call service may use the VoIP or PBXenabler, and an emergency response application may use the phone callservice, an Instant Messenger service, a video call service, and emailservice and/or a conference service.

The application layer 38 may include a conference services application63 that, together with the conference services enabler 94, enablesmultiple communication devices (including desk telephones and personalcomputers) to participate in a conference call through use of acentralized conference server 55. As seen in FIG. 5, the conferenceserver 55 is provided in the enterprise network 20 and is incommunication with the conference services enabler 94 preferably throughthe SIP protocol 80, although it is recognized that additional protocolsthat control media separate from data may be appropriate, such as theWeb Services protocol 82 or the CTI protocol 86. As will be described infurther detail below, the conference call server 55 is configured fordirecting media and data streams to and from one or more communicationdevices (i.e., mobile devices 11, telephones 17, and computers 15).

Turning now to FIGS. 6A through 7B, the general operation of the system10 using SIP 80 as the signaling protocol will be discussed, although itis recognized that the present system is not limited to the processesdiscussed herein. The signaling descriptions that follow are based onThird Party Call Control architecture, such as that illustrated in FIG.3 or 5. It will be appreciated that similar but slightly modifiedsignaling may be used in a First Party Call Control architecture,wherein the PBX 16 will pass media through to the SMP 18 for directmedia handling by the SMP 18. Variations in the signaling to adapt tovarious architectures will be appreciated by those ordinarily skilled inthe art.

FIG. 6A provides a signaling diagram for a call originating from one ofthe mobile devices 11 to a target phone 101 connected to a PrivateBranch Exchange Server or PBX 16 provided within the enterprise network20. First, the device 11 sends a mobile originated call request with itscellular number and the destination number of the target phone 101 tothe SMP 18 (block 100). In some embodiments, the mobile originated callrequest may be sent via the WLAN through the enterprise server 12. Inanother embodiment, the call request may be sent via the PLMN/PSTNthrough the PBX 16, for example as an SMS message or using anothermessaging operation. The SMP 18 confirms the call request by sending theDNIS number to the device 11 (block 102). Next, the device 11 makes acellular call using the DNIS number, which is received by the PBX 16(block 104). As the DNIS has been configured in the PBX 16 to be routedto the SMP 18 via SIP-T, in response to the incoming call, the PBX 16sends an invite over SIP-T with the DNIS number to the SMP 18 (block106). The SMP 18 matches the incoming call with the expected call fromthe mobile, and if correct, acknowledges the invite by sending a 200 OKsignal to the PBX 16, indicating that the mobile call leg is established(block 108).

The SMP 18 then sets up the outgoing call leg to the destination. Itdoes this by sending an invite over SIP-L to the PBX 16 with thedestination number of the target phone (block 110). SIP-L is used sothat the call can be correctly attributed to the individual within theorganization within any call records that are being maintained by thePBX 16. When the invite is received, the PBX 16 dials the destinationnumber to the target phone 101 (block 112), and the target phone 101answers the call (block 114). When the target phone 101 is answered, thePBX 16 sends a 200 OK signal to the SMP 18 indicating that the targetphone 101 is ready to receive data (block 115). The SMP 18 then sends aninvite over SIP-T to the PBX 16 and shuffles the SDP (SessionDescription Protocol, as known to those of ordinary skill in the art) toconnect the call legs (block 116). When the call legs are connected, thePBX 16 sends a second 200 OK signal to the SMP 18 (block 118), and theusers of the device 11 and target phone 101 can communicate with eachother.

Note that between the cellular call leg being established and theoutgoing call leg being answered, the mobile user hears ringing tones.These ringing tones may be provided by the PBX 16 using the presentationof early media from the outgoing call leg, or they may be generatedlocally on the device 11 if early media is not available. In the lattercase, it is desirable to localize the ringing tone to match the tonenormally heard with a call through the PBX 16.

The above description is known as a “mobile initiated” call, because theSMP 18 provides the mobile device 11 with the DNIS number into which themobile device 11 has called. Alternatively, the mobile originated callcould be “PBX initiated”, as shown in FIG. 6B. Specifically, in aPBX-initiated call, upon receipt of the mobile originated call request(block 120), the SMP 18 confirms receipt of the call to the mobiledevice 11 with an ANI number (block 122), which the mobile device usesto identify the incoming call from the PBX 16. The SMP 18 then sends aninvite over SIP-T to the PBX 16 with the cellular number of the deviceand the ANI number that is attached to the outgoing call (block 124).Upon receipt of the invite, the PBX 16 makes a cellular call to thedevice 11 (block 126), which is answered by the device (block 128). Thedevice 11 checks the ANI number in the incoming call to confirm if thenumber is actually from the PBX 16. If the ANI number is stripped forany particular reason, then the device 11 may be configured to answerthe call as a regular cellular call, or it may reject the call asunknown. When the device 11 answers the PBX-initiated call, the PBX 16sends a 200 OK signal to the SMP 18, indicating that the call leg to thedevice is established (block 130).

In response, the SMP 18 sends an invite over SIP-L with the destinationnumber of the target phone 101 to the PBX 16 (block 132). When theinvite is received at the PBX 16, the PBX dials the destination numberto the target phone 101 (block 134), the target phone 101 picks up thecall (block 136), and a 200 OK signal is sent from the PBX 16 to the SMP18 (block 138), indicating that the target phone 101 is also ready toreceive data. In response to the 200 OK, the SMP 18 sends an invite tothe PBX 16, shuffling the SDP to connect the call legs (block 140).Finally, when the call legs are connected, the PBX 16 sends a second 200OK signal to the SMP 18, and the users of the device 11 and target phone101 are able to communicate with each other.

In both instances, the SMP 18 is performing third party call control ofthe two call legs, the PBX 16 remaining in control of the call. Thedecision of whether to proceed with a mobile-initiated call or aPBX-initiated call can be set by policy. Specifically, the option toselect either mobile-initiated or PBX-initiated calls is a featureprovided in the SMP 18, and an administrator for the enterprise network20 can determine which setting to use. For example, in some cases it maybe more cost effective for the corporation to utilize PBX-initiatedcalls rather than mobile-initiated calls, and vice versa. However, it isappreciated that the system 10 is not limited to the above processes.

FIGS. 7A and 7B are signaling diagrams illustrating a mobile terminatedcall utilizing SIP 80. Specifically, and for the purposes of thisdisclosure, the target phone 101 is originating the call. Turning firstto FIG. 7A, an incoming call is made from the target phone 101 to thePBX 16 (block 150). When the call is received at the PBX 16, the PBX 16sends an invite to the SMP 18 over SIP-L (block 152).

In response to the invite, the SMP 18 sends a call request with the DNISnumber and source details to the device 11 (block 154), which isconfirmed to the SMP (block 156). In addition to confirming the call,the mobile device 11 sends a cellular call to the DNIS number at the PBX16 (block 158). Again, as the DNIS number is routed in the dialing plansto the SMP 18, upon receipt of the cellular call, the PBX 16 sends aninvite over SIP-T to the SMP 18 with the DNIS number (block 160). Inresponse to the invite, a “200 OK” signal is sent over SIP-T from theSMP 18 to the PBX 16, acknowledging that the call leg to the mobiledevice 11 is established (block 162). Finally, the initial invite (block152) is acknowledged with the “200 OK” signal with the cellular SDP, atwhich point the call legs are joined and the target phone 101 and device11 can communicate with each other on the call.

The diagram shown in FIG. 7A illustrates a “mobile-initiated” call,because, as discussed above with respect to FIGS. 6A and 6B, the SMP 18presents the mobile device 11 with the DNIS number at the PBX 16 intowhich to call. However, it is also possible to employ a “PBX-initiated”mobile terminated call, as shown in FIG. 7B, where the PBX 16 sends anincoming call to the device 11 with the ANI number of the target phone101.

Specifically, similar to the mobile initiated call described above andshown in FIG. 7A, the target phone 101 sends an incoming call to thedestination number of the device, which is received at the PBX 16 (block170). Upon receipt of the call, the PBX 16 sends an invite over SIP-L tothe SMP 18 (block 172) with the source number of the target phone 101.In response to the invite, the SMP 18 sends a call request with thesource number to the device 11 (block 174), with the ANI number thedevice should expect in the incoming call, the call request beingconfirmed by the device (block 176). At this point in the PBX-initiatedcall, the SMP 18 sends an invite over SIP-T to the PBX 16 with thecellular number and ANI number to use (block 178), prompting the PBX 16to make a cellular call to the device 11 with the ANI number (block180), prompting the device to ring. The device 11 answers the call(block 182), and a “200 OK” signal is sent from the PBX 16 to the SMP18, acknowledging that the cellular call leg to the device 11 isestablished (block 184). In response, a “200 OK” signal is also sentfrom the SMP 18 to the PBX 16, acknowledging that the call leg to thetarget phone 101 is also established (block 186). The SMP 18 shufflesthe SDP to connect the call legs, the call legs are joined, and thetarget phone 101 and device 11 can communicate with each other on thecall.

As discussed above with respect to FIGS. 6A and 6B, the SMP 18 typicallyremains in control of the signaling between the target phone 101 and themobile device 11 in both the mobile-initiated and PBX-initiated calls.Again, the decision to proceed with a mobile-initiated call or aPBX-initiated call is based on policy and may be set by a systemadministrator. In some cases, it may be more efficient or cost effectivefor the administrator to decide that PBX-initiated calls should be used,and in other cases, it may be more efficient or cost effective formobile-initiated calls to be utilized. As these policy decisions mayvary by organization and are not imperative to the scope of the presentapplication, they will not be discussed in further detail.

FIG. 7B also will be referenced below, with respect to FIG. 11, fordescribing examples of uplink error correction of DTMF tones used forcontrol commands and status information. In these examples, it can beassumed, for instance, that a data channel between device 11 and one ormore of SMP 18 and PBX 16 is unavailable during a telephone call. Insuch circumstances, device 11 may use the voice channel for thetelephone call to send DTMF tones. Such DTMF tones are susceptible tocorruption or failure of reception.

FIG. 8 depicts example components that can be used in implementing amobile transceiver device 109 according to the above description. FIG. 8depicts that a processing module 821 may be composed of a plurality ofdifferent processing elements, including one or more ASICs 822, aprogrammable processor 824, one or more co-processors 826, which eachcan be fixed function, reconfigurable or programmable, one or moredigital signal processors 828. For example, an ASIC or co-processor maybe provided for implementing graphics functionality, encryption anddecryption, audio filtering, and other such functions that often involvemany repetitive, math-intensive steps. Processing module 821 cancomprise memory to be used during processing, such as one or more cachememories 830.

Processing module 821 communicates with mass storage 840, which can becomposed of a Random Access Memory 841 and of non-volatile memory 843.Non-volatile memory 843 can be implemented with one or more of Flashmemory, PROM, EPROM, and so on. Non-volatile memory 843 can beimplemented as flash memory, ferromagnetic, phase-change memory, andother non-volatile memory technologies. Non-volatile memory 843 also canstore programs, device state, various user information, one or moreoperating systems, device configuration data, and other data that mayneed to be accessed persistently.

User input interface 810 can comprise a plurality of different sourcesof user input, such as a camera 802, a keyboard 804, a touchscreen 806,and a microphone, which can provide input to speech recognitionfunctionality 808. Processing module 821 also can receive input from aGPS receiver 868, which processes signals received from antenna 869.Processing module 821 also can use a variety of network communicationprotocols, grouped for description purposes here into a communicationmodule 837, which can include a Bluetooth communication stack 842, whichcomprises a L2CAP layer 844, a baseband 846 and a radio 848.Communications module 837 also can comprise a Wireless Local AreaNetwork (847) interface, which comprises a link layer 852 with a MAC854, and a radio 856. Communications module 837 also can comprise acellular broadband data network interface 850, which in turn comprises alink layer 861, with MAC 862. Cellular interface 850 also can comprise aradio for an appropriate frequency spectrum 864. Communications module837 also can comprise a USB interface 866, to provide wired datacommunication capability. Other wireless and wired communicationtechnologies also can be provided, and this description is exemplary.

Referring to FIG. 9, there is depicted an example of mobile device 11.Mobile device 11 comprises a display 912 and a cursor or viewpositioning device, here depicted as a trackball 914, which may serve asanother input member and is both rotational to provide selection inputsand can also be pressed in a direction generally toward housing toprovide another selection input. Trackball 914 permits multi-directionalpositioning of a selection cursor 918, such that the selection cursor918 can be moved in an upward direction, in a downward direction and, ifdesired and/or permitted, in any diagonal direction. The trackball 914is in this example situated on a front face (not separately numbered) ofa housing 920, to enable a user to maneuver the trackball 914 whileholding mobile device 11 in one hand. In other embodiments, a trackpador other navigational control device can be implemented as well.

The mobile device 11 in FIG. 9 also comprises a programmable conveniencebutton 915 to activate a selected application such as, for example, acalendar or calculator. Further, mobile device 11 can include an escapeor cancel button 916, a menu or option button 924 and a keyboard 920.Menu or option button 924 loads a menu or list of options on display 912when pressed. In this example, the escape or cancel button 916, menuoption button 924, and keyboard 920 are disposed on the front face ofthe mobile device housing, while the convenience button 915 is disposedat the side of the housing. This button placement enables a user tooperate these buttons while holding mobile device 11 in one hand. Thekeyboard 920 is, in this example, a standard QWERTY keyboard.

FIG. 10 depicts an example functional module organization of mobiledevice 11. Call module 1001 identifies a logical organization of moduleswhich can be used for implementing aspects described herein.

The FIG. 10 example of device 11 also depicts a speech codec 1010, whichcan do one or more of coding and decoding speech obtained or transmittedon the voice channel and a tone injection module 1008. Speech coder 1010and tone injection module 1008 both can provide inputs to a voicechannel processing layer 1018. Both data channel processing layer 1016and voice channel processing layer 1018 can send and receive data to andfrom transport protocol(s) layer 1020, which in turn communicates withMAC/PHY 1022.

FIG. 11 depicts a mobile device 11 that can communicate over a voicechannel 1105 with PBX 16, which in an example can comprise a DTMF tonematching module 1102 that finds matches for tones that are detected onvoice channel 1105. Each device 11 and PBX 16 can have access todescription for DTMF codes that match to given commands or statusindicators, and which can be stored on a computer readable medium,represented as feature codes 1116 in FIG. 11. The tones that are definedto indicate such commands or status indicators are used in comparisonswith the tones that received on voice channel 1105, and which ultimatelycan output a command 1108 (generic to command or status information orother information to be communicated). FIG. 11 depicts that feature codeA44A was transmitted by device 11 on voice channel 1105.

FIG. 12 depicts an example composition of tone matching module 1102. Inone example, tone matching module 1102 can comprise a DTMF tone detector1120, which monitors voice channel 1105 and outputs indicators of DTMFtones that it detects. For example, for the A44 code transmitted, tonedetector 1120 can output 4A, A4, or A4A (not necessarily an exhaustivelist, but for explanation). In other words, some tones can be lost ornot be detected by detector 1120, for any of a variety of reasons. Forexample, the tones can fail to be detected because SDP ports were beingshuffled during a call transfer.

The tones recognized tone are provided to a compare module 1122, whichcompares the tones provided from detect module 1120 to the tones thatrepresent each feature code. In this example, if tones A4 were detected,then those tones can be matched to a start delimiter (A), and a firstinformational tone (4). If 4A was received, then the informational tonereceived (4) can be matched to the last informational tone of thedefinition, and the delimiter can be matched to the ending delimiter. Assuch, a code can be reconstructed in the absence of DTMF digit loss.Similarly, if one of the informational tones is lost (either 4), thenthe received informational tone can be matched to either tone, given thereception of the start and stop delimiter tones. It is preferred thatmore loss prone situations use redundant informational tones. Forexample, a command from device 11 to cancel an in-progress transferpreferably is assigned a code that has two or more repeatinginformational tones.

The tone combination that is determined by compare module 1122 can beprovided to a code matching module 1124, which outputs a command/code1108 that is indicative of the command or status desired to beindicated.

FIG. 13 depicts that compare module 1122 can employ state-dependentanalysis techniques. For example, at call state 1305, a next state isproceed 1306 or fail 1307. The code to be received is A44A to advance toproceed 1306. Thus, if a code similar to A44A comes in during that time,compare module 1122 can select proceed 1306. At other times, if A44A isnot a code that advances to another available state, then A44A would beless likely to be outputted by compare module 1122.

In this description, tones A, B, C, and D may be referenced, which aredefined respectively as a combination of (1) a 1633 Hz tone and (2) asecond tone at 697 Hz (for A), 770 Hz (for B), 852 Hz (for C), and 941Hz (for D). It may be the case that some networks do not support some orall of these tones, and as such, although these tones can be usedpreferentially as delimiter and/or informational tones, if there is adetermination that such tones are not supported for a given network (canbe based on network baseband technology, such as GSM versus CDMA), thenother DTMF tones can be used. Of course, DTMF tones can be synthesizedas well, which are not a priori assigned to a given digit on the keypad,if a given network and device supports such functionality.

FIG. 14, which depicts a method in which device 11 can participate,includes establishing a voice channel for a call (block 1402). Then,reception of a command from a UI can be monitored (block 1404). If acommand is received (e.g., transfer, or cancel), then it can bedetermined whether a data channel is available (block 1406). If a datachannel is available, device 11 can signal (block 1408) the command overthe data channel. If there is no data channel available, then device 11can fail over to using the voice channel for command transfer. For usingthe voice channel the command is translated into a DTMF sequence (e.g.,by looking the command up to find a matching sequence from the storedfeature code data 1116 (block 1410). The DTMF sequence is sent over thevoice channel (block 1413). The method further comprises continuing tomonitor for additional commands (block 1414), and returning totranslation upon detecting such commands. In absence of such detection,the method can wait (block 1416) and monitor. For ease of explanation,the term command was used but more generally, status information,commands, or other information can be transferred according to thisdisclosure.

FIG. 15 depicts a method that can be implemented by a server or PBX 18in receiving the tones and determining what information is indicatedthereby. The method includes monitoring (block 1502) the voice channelto detect a delimiter tone (block 1504). If a delimiter tone is detectedthen the method waits for detection of an informational tone, and if aninformation tone is detected (block 1506), the method loops to detectanother. If however, the informational tone is not detected, a delimitertone may be detected (block 1510), which would be the stop delimiter ofthe tones shown as the feature codes of 1116. Upon reception of suchdelimiter, the received tones can be translated (1514) into a code (fromthe list of 1116, for example), and a command can be determined (1516)from the DTMF code determined. The command can be outputted (1520). If adelimiter tone was not detected at block 1510, then a timeout can besensed (block 1511), and if there was a timeout, then translation (block1514) can occur with what tones were received. If the timeout did notoccur then detection of any of informational tones and delimiter tonescan continue, absent reception of the delimiter at block 1510 (i.e., adelimiter received after reception of either a first delimiter tone orat least one informational tone). These figures depict exampleapproaches; however, other implementations are possible that remainlogically equivalent to these examples.

In the foregoing, separate boxes or illustrated separation of functionalelements of illustrated systems does not necessarily require physicalseparation of such functions, as communications between such elementscan occur by way of messaging, function calls, shared memory space, andso on, without any such physical separation. As such, functions need notbe implemented in physically or logically separated platforms, althoughthey are illustrated separately for ease of explanation herein.

For example, different embodiments of devices can provide some functionsin an operating system installation that are provided at an applicationlayer or in a middle layer in other devices. Different devices can havedifferent designs, such that while some devices implement some functionsin fixed function hardware, other devices can implement such functionsin a programmable processor with code obtained from a computer readablemedium.

Further, some aspects may be disclosed with respect to only certainexamples. However, such disclosures are not to be implied as requiringthat such aspects be used only in embodiments according to suchexamples.

The above description occasionally describes relative timing of events,signals, actions, and the like as occurring “when” another event,signal, action, or the like happens. Such description is not to beconstrued as requiring a concurrency or any absolute timing, unlessotherwise indicated.

Certain adaptations and modifications of the described embodiments canbe made. Aspects that can be applied to various embodiments may havebeen described with respect to only a portion of those embodiments, forsake of clarity. However, it is to be understood that these aspects canbe provided in or applied to other embodiments as well. Therefore, theabove discussed embodiments are considered to be illustrative and notrestrictive.

What is claimed is:
 1. A method of receiving commands during a call overa channel, the commands based on tone description data for at least onefeature code, each at least one feature code defined by a startdelimiter tone, a stop delimiter tone, and a pre-determined number of atleast two informational tones, the method comprising: receiving a signalover the channel; identifying, in the received signal, a delimiter toneand at least one informational tone, but fewer than the pre-determinednumber of informational tones, when one or more of the at least twoinformational tones is not detected in the received signal; anddetermining a feature code based on the identified delimiter tone andthe identified at least one informational tone.
 2. The method of claim1, wherein the determining comprises using a state of the call indetermining candidate feature codes for commands useable during thatcall state.
 3. The method of claim 2, wherein the state of the callcomprises a call transfer.
 4. The method of claim 3, wherein thedetermined feature code is indicative of a cancellation of the calltransfer.
 5. The method of claim 2, wherein the state of the callcomprises that the call has been connected, and the at least oneinformational tone comprises one tone which is matched to anacknowledgment feature code.
 6. The method of claim 1, wherein the atleast one feature code comprises a cancel transfer code, which isdefined as a start delimiter tone comprising a combination of (1) a 1633Hz tone and (2) two repeating DTMF tones and (3) an ending delimitertone comprising a combination of (1) a 1633 Hz tone and (2) a toneselected from the set consisting of about 697 Hz, 770 Hz, 852 Hz, and941 Hz.
 7. The method of claim 1, wherein the identifying of thedelimiter tone comprises identifying either the start delimiter tone orthe stop delimiter tone, and the identifying of the at least oneinformational tone comprises identifying one of two or more repeatingDTMF tones.
 8. The method of claim 1, wherein the identifying of thedelimiter tone comprises identifying the starting delimiter tone and theending delimiter tone, and the identifying of the at least oneinformational tone comprises identifying exactly one or exactly two oftwo or more available repeating DTMF tones, and the determiningcomprises matching the sequence of the start delimiter tone, the atleast one informational tone, and the stop delimiter tone to acorresponding feature code.
 9. The method of claim 1, wherein the atleast one feature code comprises a finish transfer code, which isdefined as a start delimiter tone comprising a combination of (1) a 1633Hz tone and (2) two repeating DTMF tones and (3) a stop delimiter tonecomprising a combination of (1) a 1633 Hz tone and (2) a tone selectedfrom the set consisting of about 697 Hz, 770 Hz, 852 Hz, and 941 Hz. 10.A system for providing enterprise services, comprising: an electronicdevice comprising: a channel interface, a processor coupled to theinterface, and a non-transitory computer readable medium storinginstructions executable by the processor to perform a method comprising:receiving a control command through the interface, mapping the controlcommand to a sequence descriptive of DTMF tones comprising a startdelimiter, at least two informational tones, and a stop delimiter, andcausing the sequence descriptive of DTMF tones to be modulated via thechannel interface; and a server comprising: a first channel interfaceconfigurable to communicate over a channel with the channel interface ofthe electronic device, a non-transitory computer readable medium storingsequences descriptive of DTMF tones composing each of a group of featurecodes, a detection module operable to detect tones on the first channelinterface, and a translation module configurable for mapping a detecteddelimiter tone and at least one detected informational tone, when one ormore of the at least two informational tones is not detected, into oneof the group of feature codes by matching the composition either to thestart delimiter and an informational tone that follows or the stopdelimiter and a preceding informational tone.
 11. The system of claim10, wherein the translation module is further configurable to map basedon a current call state.
 12. The system of claim 10, wherein thetranslation module is operative to map the detected delimiter tone andthe detected informational tone to at least one tone corresponding to atleast one feature code available during a current call state.
 13. Amethod comprising: determining, at an electronic device, a command forcontrolling a voice telephone call between the electronic device and aterminating entity; formulating a DTMF tone sequence indicative of afeature code, the DTMF tone sequence comprising a start delimiter tone,at least two informational tones such that a feature code isdeterminable by the terminating entity even if one or more of the atleast two information tones is not detected, and a stop delimiter tone,each of the start delimiter tone and the stop delimiter tone comprisinga DTMF tone defined by a combination of a 1633 Hz tone and a toneselected from the set consisting of about 697 Hz, 770 Hz, 852 Hz, and941 Hz; and modulating the DTMF tone sequence on a voice channel betweenthe electronic device and the terminating entity.
 14. The method ofclaim 13, wherein the control command comprises a cancel of a transferoperation command, and the at least two informational tones comprise atleast two repeating DTMF tones.
 15. A computer-implemented method ofreceiving commands during an in-progress call over a voice channel, thecommands based on tone description data for a group of feature codes,each feature code of the group respectively defined by a start delimitertone, a stop delimiter tone, and a pre-determined number ofinformational tones, the method comprising: receiving voice band signalsover the voice channel established for the in-progress voice callbetween a mobile device and a terminating entity; identifying, in thereceived voice band signals, a delimiter tone and at least oneinformational tone but fewer than the pre-determined number ofinformational tones; determining a feature code from the group offeature codes based on the received delimiter tone and the receivedinformational tones; and outputting the determined feature code, whereinthe group of feature codes comprises a cancel transfer code, which isdefined as a starting delimiter tone comprising a combination of (1) a1633 Hz tone and (2) two repeating DTMF tones and (3) an endingdelimiter tone comprising a combination of (1) a 1633 Hz tone and (2) atone selected from the set consisting of about 697 Hz, 770 Hz, 852 Hz,and 941 Hz.
 16. The method of claim 15, wherein the identifying, in thereceived voice band signals, of the delimiter tone comprises identifyingeither the starting delimiter tone or the ending delimiter tone, and theidentifying of the at least one informational tone comprises identifyingone of two or more repeating DTMF tones, and the determining comprisesmatching the sequence of the identified delimiter and the one identifiedinformational tone to corresponding tones of the cancel transfer code.17. The method of claim 15, wherein the identifying, in the receivedvoice band signals, of the delimiter tone comprises identifying thestarting delimiter tone and the ending delimiter tone, and theidentifying of the at least one informational tone comprises identifyingonly one of two or more available repeating DTMF tones, and thedetermining comprises matching the sequence of the starting delimitertone, the one informational tone, and the ending delimiter tone to acorresponding feature code.
 18. A computer-implemented method ofreceiving commands during an in-progress call over a voice channel, thecommands based on tone description data for a group of feature codes,each feature code of the group respectively defined by a start delimitertone, a stop delimiter tone, and a pre-determined number ofinformational tones, the method comprising: receiving voice band signalsover the voice channel established for the in-progress voice callbetween a mobile device and a terminating entity; identifying, in thereceived voice band signals, a received delimiter tone and at least onereceived informational tone but fewer than the pre-determined number ofinformational tones; determining a feature code from the group offeature codes based on the received delimiter tone and the receivedinformational tones; and outputting the determined feature code, whereinthe group of feature codes comprises a finish transfer codecorresponding to a starting delimiter tone comprising a combination of(1) a 1633 Hz tone and (2) two repeating DTMF tones and (3) an endingdelimiter tone comprising a combination of (1) a 1633 Hz tone and (2) atone selected from the set consisting of about 697 Hz, 770 Hz, 852 Hz,and 941 Hz.
 19. The method of claim 18, wherein the identifying, in thereceived voice band signals, of the delimiter tone comprises identifyingeither the starting delimiter tone or the ending delimiter tone, and theidentifying of the at least one received informational tone comprisesidentifying one of two or more repeating DTMF tones, and the determiningcomprises matching the sequence of the identified delimiter and the oneidentified informational tone to corresponding tones of the canceltransfer code.
 20. The method of claim 18, wherein the identifying, inthe received voice band signals, of the delimiter tone comprisesidentifying the starting delimiter tone and the ending delimiter tone,and the identifying of the at least one informational tone comprisesidentifying only one of two or more available repeating DTMF tones, andthe determining comprises matching the sequence of the startingdelimiter tone, the one informational tone, and the ending delimitertone to a corresponding feature code.